U.S. patent number 7,251,254 [Application Number 10/653,772] was granted by the patent office on 2007-07-31 for telecommunication network system and method in communication services using session initiation protocol.
This patent grant is currently assigned to AT&T Corp.. Invention is credited to Gregory W. Bond, Eric Cheung, Kermit Hal Purdy, Xiaotao Wu, Pamela Zave.
United States Patent |
7,251,254 |
Bond , et al. |
July 31, 2007 |
Telecommunication network system and method in communication
services using session initiation protocol
Abstract
An implementation of a voice-over-Internet protocol (VoIP)
system for accomplishing two-way, three-way and conference calling
between two or more parties is disclosed, in which new call
features are readily adapted. The VoIP system is implemented in a
Session Initiation Protocol (SIP) framework in which aspects of
Distributed Feature Composition (DFC) architecture are modified and
applied to overcome known limitations in the adaptability of
existing VoIP frameworks.
Inventors: |
Bond; Gregory W. (Hoboken,
NJ), Cheung; Eric (New York, NY), Purdy; Kermit Hal
(Bernardsville, NJ), Wu; Xiaotao (New York, NY), Zave;
Pamela (Chatham, NJ) |
Assignee: |
AT&T Corp. (New York,
NY)
|
Family
ID: |
34217970 |
Appl.
No.: |
10/653,772 |
Filed: |
September 3, 2003 |
Prior Publication Data
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|
|
|
Document
Identifier |
Publication Date |
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US 20050047389 A1 |
Mar 3, 2005 |
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Current U.S.
Class: |
370/467;
370/466 |
Current CPC
Class: |
H04L
29/06027 (20130101); H04L 63/08 (20130101); H04L
65/403 (20130101); H04L 65/1006 (20130101) |
Current International
Class: |
H04J
3/16 (20060101) |
Field of
Search: |
;370/466,467,400,401 |
References Cited
[Referenced By]
U.S. Patent Documents
Primary Examiner: Smith; Creighton
Claims
What is claimed is:
1. A method for establishing a call feature for a network
telecommunications call, comprising: receiving an incoming session
initiation protocol (SIP) request; determining a feature to be
applied based on a parameter of the SIP request; assigning a
feature box corresponding to the feature; converting at least a
portion of the SIP request to a distributed feature composition
(DFC) signaling message; and routing the request to the feature box
using the DFC signaling message.
2. The method of claim 1, said incoming SIP request comprising a
call request from a caller to a callee.
3. The method of claim 2, said parameter comprising at least one
of: a source address of the caller and a target address of the
callee.
4. The method of claim 3, wherein at least one of the caller and
the callee is a subscriber of the feature to be applied.
5. The method of claim 2, said feature comprising at least one of:
a call forwarding feature, a three-way calling feature, and a call
conferencing feature.
6. The method of claim 2, said feature including a follow-me call
feature in which a new target address for the callee is applied to
the call request.
7. The method of claim 6, said follow-me call feature is
implemented by at least two follow-me feature boxes.
8. The method of claim 7 further comprising: determining a priority
of target addresses for the callee using the at least two follow-me
feature boxes.
9. The method of claim 2, further comprising: converting the DFC
signaling message to an outgoing SIP request after said routing;
and routing the outgoing SIP request to a target address
corresponding to the callee.
10. The method of claim 2, said DFC signaling message comprising an
identification of a signaling path and a media path for the
incoming SIP request.
11. The method of claim 10, said signaling path comprising an
internal signaling path including the feature box and said media
path comprising an external path for a media flow corresponding to
the call request.
12. The method of claim 1, said incoming SIP request comprising a
voice over Internet protocol (VoIP) call request.
13. The method of claim 1, said incoming SIP request comprising a
registration request, said feature box comprising an SIP
registration feature box.
14. The method of claim 1, said determining comprising: querying a
database for the feature to be applied to the request.
15. The method of claim 1, said converting further comprising:
storing the parameter as an SIP parameter in at least one field of
the DFC signaling message.
16. The method of claim 1, said routing further comprising: routing
the request by a featureless call to the feature box, wherein the
feature box is dedicated to the request.
17. The method of claim 1, said feature comprising a plurality of
features to be assigned to the request.
18. The method of claim 17, said routing further comprising:
routing the request to a plurality of feature boxes in a
predetermined order, each feature box assigned to one of the
plurality of features.
19. The method of claim 18, said routing further comprising:
routing the request to a first of the plurality of feature boxes
based on the predetermined order of the plurality of feature
boxes.
20. The method of claim 18, each said feature box corresponding to
assigned zones of feature boxes, said predetermined order
corresponding to a first priority of the assigned zones, and a
second priority of features to be assigned within each of the
assigned zones.
21. The method of claim 20, said assigned zones comprising a source
zone, a dialed zone and a target zone.
22. The method of claim 1, further comprising: receiving a new
feature for the request; and rerouting the request to a second
feature box assigned to the new feature.
23. A method for routing a telecommunications request, comprising:
receiving a request including a DFC signaling message having an SIP
parameter, and routing the request to a plurality of feature boxes
in a predetermined order, each feature box corresponding to a
feature assigned to the request.
24. The method of claim 23, said request comprising a call request
from a caller to a callee.
25. The method of claim 24, wherein at least one of the caller and
the callee is a subscriber of the feature assigned to the
request.
26. The method of claim 24, said feature comprising at least one
of: a call forwarding feature, a three-way calling feature, a
follow-me feature and a call conferencing feature.
27. The method of claim 24, further comprising: routing the request
to the callee after the request has been routed through a last of
the plurality feature boxes.
28. The method of claim 23, each said feature box corresponding to
assigned zones of feature boxes, said predetermined order
corresponding to a first priority of the assigned zones, and a
second priority of features to be assigned within each of the
assigned zone.
29. A method for completing a call request to a callee, comprising:
receiving a DFC signaling message from a last of a plurality of
feature boxes, the DFC signaling message corresponding to a call
request from a caller to a callee; converting the DFC signaling
message to an SIP request; and transmitting the SIP request to a
target address of the callee.
30. The method of claim 29, said converting further comprising:
retrieving an SIP parameter for the call request from at least one
field of the DFC signaling message; and generating the SIP request
from the SIP parameter.
31. The method of claim 29, further comprising: determining the
target address based on a priority of callee addresses in
accordance with a follow-me feature to which the callee is
subscribed.
32. A method for routing a call request, comprising: receiving an
incoming SIP call request from a caller to a callee, the callee
comprising a subscriber to a follow-me feature; routing the call
request to a follow-me feature box using a DFC signaling message
corresponding to the incoming SIP call request; determining a
priority of target addresses for the callee in accordance with
stored callee preferences; and generating an outgoing SIP request
for establishing a call between the caller and the callee, the
outgoing SIP request transmitted sequentially to each target
address according to the priority until the call is answered by the
callee.
33. The method of claim 32, said follow-me feature box comprising
an SIP follow-me feature box for handling SIP follow-me parameters
and a general follow-me feature box for handling third-party
follow-me parameters.
34. The method of claim 33, the general follow-me feature box
having a higher priority in the routing of the call than the SIP
follow me feature box.
35. An apparatus for establishing a call feature for a network
telecommunications call, comprising: means for receiving an
incoming session initiation protocol (SIP) request; means for
determining a feature to be applied based on a parameter of the SIP
request; means for assigning a feature box corresponding to the
feature; means for converting at least a portion of the SIP request
to a distributed feature composition (DFC) signaling message; and
means for routing the request to the feature box using the DFC
signaling message.
36. An apparatus for establishing a call feature for a network
telecommunications call, comprising: a processor; and a memory in
communication with the processor, the memory storing a plurality of
processing instructions directing the processor to: receive an
incoming session initiation protocol (SIP) request; determine a
feature to be applied based on a parameter of the SIP request;
assign a feature box corresponding to the feature; convert at least
a portion of the SIP request to a distributed feature composition
(DFC) signaling message; and route the request to the feature box
using the DFC signaling message.
Description
FIELD OF THE INVENTION
The present disclosure relates generally to telephonic
communications, and relates more particularly to Internet-based
telecommunications multimedia communications.
BACKGROUND OF THE INVENTION
The Voice over Internet Protocol (VoIP) community has been
developing and improving voice communications systems and
applications implemented over the Internet, with the goal of
providing, improving upon and expanding call services that are
commonly available over public switched telephone networks (PSTNs)
alone. Session Initiation Protocol (SIP) is one existing
application-layer VoIP for creating, modifying, and terminating
call sessions involving one or more users. Such sessions may
include Internet telephone dialogs and sessions, multimedia
distribution, multimedia conferences, and the like. In existing SIP
domains, distributed proxy servers route requests to a user's
current location, authenticate and authorize users for various
services, implement provider call-routing policies, and provide
various call features to users. Call invitation requests are used
to create sessions, and carry session descriptions that allow
callers and callees to agree on a set of compatible media types for
accomplishing multi-party communications. SIP also provides a
registration function for a user to subscribe to various call
features, such as three-way calling. SIP registration functions
also allow users to upload their current locations for use by proxy
servers that receive such call requests for a user.
Similar to what has been encountered during the evolution of PSTNs,
as new options are added to the suite of available call features in
SIP, it becomes increasingly difficult to manage the behavioral
complexity of new call features and their resulting interactions
with existing call features.
Distributed feature composition (DFC) has been previously developed
by certain of the inventors of the present application to contain
this feature-interaction problem in PSTN environments. A
description of DFC may be found in U.S. Pat. Nos. 6,404,878 and
6,160,883, both of which are assigned to the assignee of the
present invention and incorporated herein by reference. DFC allows
for the modification and addition of call features while minimizing
unexpected call feature interference and system malfunctions. It is
a virtual architecture within which a call feature is implemented
by a small number of physical components, collectively called a
feature box, that are added to a call route by featureless internal
calls through the communications network and are connected by known
PSTN network mechanisms. A customer call is handled by building a
configuration of feature boxes that are dedicated to the
appropriate call features assigned to the call in a predetermined
order of priority.
Any new desired features are implemented in DFC by as small a
number of new components as possible, preferably just one, along
with predetermined rules for joining the new components. The
architectural style of DFC is similar to dynamic pipe-and-filter
technologies, where the feature boxes are like filters, and the
internal featureless call connections between feature boxes are
like pipes. DFC feature boxes are physically independent entities,
each with its own state, sharing no state with other feature boxes
and independent of the identities of its neighboring components.
DFC feature boxes are also completely dedicated to an assigned
call, and only become available to other calls when the assigned
call is terminated or cancelled. This independence contributes
significantly to the ability of DFC to manage feature interactions,
however, requires a large number of physical components to
accommodate a large network of users.
DFC architecture may not simply be incorporated into SIP
frameworks. First, the systems are implemented in different domain
environments having different component types and signaling
protocols. The requirements for certain DFC and SIP hardware also
conflict in certain areas of implementation. However, it would be
beneficial if SIP could be improved to allow for the ready
accommodation of new call features therein.
Third party service developments, such as Call Processing Language
and SIP Common Gateway Interface, as well as a variety of
distributed component architectures and feature interaction
protocols, such as Software Engineering Research Library (SERL),
Third Party Call Control (3PCC) and Application Server Component
(ASC), have been proposed to improve VoIP performance. However,
none has definitively addressed the feature interaction problem in
SIP.
SUMMARY OF THE INVENTION
It is an object of the present disclosure, therefore, to have a
system directed to particular components of an improved VoIP
communications protocol that is readily adaptable to new call
features. In particular, one aspect of the invention includes a
method and apparatus for establishing one or more call features
within a VoIP domain for handling a network telecommunications
call. The domain has a predetermined protocol for handling call
invite and other call requests. In certain embodiments, an incoming
SIP request is translated to an internal DFC signaling message that
includes an identification of a signaling path and a media path for
the call request. The internal routing of the call to appropriate
feature boxes may be accomplished using such DFC signaling
messages. Existing SIP parameters in the call request may remain
unchanged in the translation. Feature boxes are assigned to the
call request based on its assigned call features and the
predetermined priority of call features.
In various embodiments, a feature box is not a physically separate
physical component, but may be implemented as dedicated logical
unit of a telecommunications server or the like. Individual feature
boxes are dedicated to a call request until a call is terminated. A
follow-me call feature, in which a new address for a call request
to a user is established in accordance with user customizable
preferences, may be implemented by one or more separate follow-me
feature boxes.
In still further embodiments, the media flow corresponding to the
call request may be routed directly between two endpoints of the
call, and the route may be exclusive of any feature box for a
featureless call.
BRIEF DESCRIPTION OF THE DRAWINGS
Further aspects of the present disclosure will be more readily
appreciated upon review of the following detailed description of
its various embodiments when taken in conjunction with the
accompanying drawings, of which:
FIG. 1 is a schematic diagram of an exemplary SIP network
environment for the deployment of feature boxes;
FIG. 2 is a schematic diagram of an exemplary domain within the SIP
network environment of FIG. 1,
FIG. 3 is a schematic diagram of call flow through the feature
boxes of the domain of FIG. 2;
FIG. 4 is a call flow chart of an exemplary successful call setup
process in the SIP network environment of FIG. 1;
FIG. 5 is a call flow chart of an exemplary failed call setup
process in the SIP network environment of FIG. 1; and
FIG. 6 is a call flow chart of exemplary registration processes in
the SIP network environment of FIG. 1.
DETAILED DESCRIPTION OF THE SPECIFIC EMBODIMENTS
Referring now to FIGS. 1-6, wherein similar components of the
present disclosure are referenced in like manner, various
embodiments of a telecommunication network system and method in
communication services using Session Initiation Protocol are
disclosed. The existing structure for SIP and DFC systems will now
be described in greater detail to illustrate the distinctions
between prior schema and those of the present disclosure, as well
as prior elements that may be used to enable certain functions
described herein.
Previous implementations of SIP are structured as a layered
protocol in which a set of generally independent processing stages
are coupled together. This description of protocol behavior as
layers is done for purposes of illustration only. It is not meant
in any way to dictate a necessary feature for implementation. Each
layer is compliant to a set of predetermined rules defined by that
layer. Not every physical component specified by the SIP protocol
utilizes every layer. Furthermore, the elements specified by SIP
are logical elements, not physical ones, although a physical
realization could be accomplished with physically distinct logical
hardware elements and on a transaction-by-transaction basis. The
existing SIP environment will be briefly described with respect to
the types of hardware employed and the format for communications
among these components. A complete description of SIP can be found
at IETF.ORG.
A first program layer of SIP is its syntax and encoding layer.
Encoding is specified using an augmented Backus-Naur Form (BNF)
grammar having a standard message structure.
A second layer is the transport layer that defines how clients and
servers send or receive requests and responses over the network.
All SIP elements contain the transport layer.
A third layer is the transaction layer. A transaction is a call
request sent by a client to a server, along with all responses to
that request sent from the server transaction back to the client,
using the transport layer. The transaction layer encompasses client
transactions and server transactions, each of which are performed
by an SIP component, such as a finite state machine (FSM) that is
dedicated to process a particular request. The transaction layer
handles application-layer retransmissions, matching of responses to
requests, and application-layer timeouts. Any task that a user
agent (UA) accomplishes, as described further below, may take place
using a series of client and server transactions. In SIP, UAs
contain a transaction layer, as do SIP stateful proxies. Stateless
proxies, however, do not contain a transaction layer.
A fourth layer is the transaction user (TU) layer. Each of the SIP
entities, except the stateless proxy, is designated as a TU. When a
client TU wishes to send a call request, it creates a client
transaction instance and passes the request with the destination IP
address, port, and transport.
A TU that creates a client transaction may also cancel it. When a
client cancels a transaction, it requests that the server stop
further processing, revert to the state that existed before the
transaction was initiated, and generate a specific error response
to that transaction. This is done with a CANCEL request, which
constitutes its own transaction.
SIP components, including UA clients and servers, stateless and
stateful proxies and registrars, contain a core that distinguishes
them from each other. Cores, except for the stateless proxy, act as
TUs within the environment and employ the TU layer. While the
behavior of UA client and server cores depends on the particular
circumstances of a transaction, there are some common rules
applicable to both. For a UAC, these rules govern the construction
of a call request, and for a UA server generally, they govern the
processing of call requests and generating responses thereto.
Registration of users plays an important role in SIP applications.
A dedicated UA server, referred to as an SIP registrar, is thus
designated to handle REGISTER requests.
In order to avoid malicious floods of unauthenticated requests,
known as denial-of-service attacks, a stateless UA is used to
handle unauthenticated requests for which a challenge response is
to be issued. The stateless UA replies to requests normally, but
discards any state that would ordinarily be retained by a UA server
after a response has been sent. If a stateless UA server receives a
retransmission of a request, it regenerates the response and
resends it, just as if it were replying to the first instance of
the request. Stateless UA servers do not use a transaction layer.
Instead, they receive requests directly from the transport layer
and send responses directly to the transport layer.
A back-to-back user agent (B2BUA) is a logical entity that receives
a call request and processes it as a UA server. In order to
determine how the request should be answered, it also acts as a UA
client and generates requests. Unlike a proxy server, it maintains
dialog state and must participate in all requests sent on the
dialogs it has established.
The most important method in SIP is the INVITE method, which is
used to establish a session or dialog between users. A dialog is a
peer-to-peer SIP relationship between two user agents that persists
for some time. The dialog facilitates sequencing of messages and
proper routing of requests between user agents. A session is a
collection of participants, and the streams of media flow
there-between. A session may include one or more SIP dialogs.
A status-code is a 3-digit integer that is generated in SIP to
indicate the outcome of an attempt to respond to a call request.
The 3-digit system has similarities to that used in a hyper-text
transfer protocol (HTTP) environment. The first digit of the
status-code is assigned to a particular class of response. The last
two digits have only a sequential role in identifying individual
responses of a particular class. Any response with a status-code
between 100 and 199 is referred to as a "1xx response", any
response with a status code between 200 and 299 as a "2xx
response", and so on. SIP allows six values for the first digit,
with classes assigned to each of digits 1-6 as follows:
1xx: Provisional responses indicating that a request was received,
and that a UA is continuing to process the request.
2xx: Success response classes in which the call request was
successfully received, understood, and accepted.
3xx: Redirection responses indicating that further action needs to
be taken in order to complete the call request.
4xx: Client Error responses in which the request contains bad
syntax or cannot be fulfilled by a UA.
5xx: Server Error responses in which the UA fails to fulfill an
apparently valid request.
6xx: Global Failure responses in which the request cannot be
fulfilled at any server.
A reason-phrase is intended to give a short textual description of
the status-code to a human operator or user. Each status-code also
dictates an appropriate action to take in response to the received
code (which is a default action unless otherwise indicated). It
should be readily apparent that no SIP component is required to
examine or display any reason-phrase. The following chart of
specific wording for the reason-phrase may be altered or omitted in
various embodiments:
TABLE-US-00001 STATUS- CODE REASON-PHRASE ACTION 100 TRYING stops
retransmission of INVITE 180 RINGING multiple status messages sent
upstream 183 SESSION PROGRESS header fields, or message body may be
used to convey more details about the call progress. 200 OK 3xx
WARNING CODE 400 BAD REQUEST 401 UNAUTHORIZED 402 PAYMENT REQUIRED
403 FORBIDDEN 404 NOT FOUND send unknown status message followed by
teardown 405 METHOD NOT ALLOWED 406 NOT ACCEPTABLE 407 PROXY
AUTHENTICATION REQUIRED 408 REQUEST TIMEOUT 409 CONFLICT 410 GONE
413 REQUEST ENTITY TOO LARGE 414 REQUEST-URI TOO LONG 415
UNSUPPORTED MEDIA TYPE 420 BAD EXTENSION 480 TEMPROARILY
UNAVAILABLE 481 CALL LEG/TRANSACTION DOES NOT EXIST 482 LOOP
DETECTED 483 TOO MANY HOPS 484 ADDRESS INCOMPLETE 485 AMBIGUOUS 486
BUSY HERE 487 REQUEST TERMINATED 488 NOT ACCEPTABLE HERE 500 SERVER
INTERNAL ERROR 501 NOT IMPLEMENTED 502 BAD GATEWAY 503 SERVICE
UNAVAILABLE 504 SERVER TIME-OUT 505 VERSION NOT SUPPORTED 513
MESSAGE TOO LARGE 600 BUSY EVERYWHERE 603 DECLINE sends reject
message on all channels followed by teardown 604 DOES NOT EXIST
ANYWHERE send unknown status message followed by teardown 606 NOT
ACCEPTABLE
SIP header fields are similar to HTTP header fields in both syntax
and semantics. In particular, SIP header fields follow the
definitions of syntax for the message header and the rules for
extending header fields over multiple lines. Header fields which
are needed for proxy processing, such as via, route, record-route,
proxy-require, max-forwards, and proxy-authorization, are placed at
the top of an SIP request to facilitate rapid parsing.
The details of a session or dialog, such as the type of media,
codec, or sampling rate, are not described using SIP. Rather, the
body of a SIP message contains a description of the session that is
generally encoded in some other protocol format, such as Session
Description Protocol (SDP) and included in an SIP message.
Existing implementations of DFC are meant for deployment primarily
over PSTNs and VoIP domains. The routing of calls between feature
boxes and/or other DFC components is the responsibility of a
router. A user's request for service typically causes the caller's
trunk interface (TI) box to send a setup message to a DFC router. A
setup message is a request to create an internal call. The router
routes the call to a feature box, which receives it and then
completes the protocol to set up the internal call by connecting
the interface box and the feature box. Typically a feature box that
receives an incoming internal call places a corresponding outgoing
internal call. The outgoing call is placed using the setup message
received as part of the incoming call, and is considered a
continuation of the incoming call. This creates a chain of feature
boxes and internal calls, which extends through all the applicable
feature boxes to a TI box of a final target address for the
call.
A chain of feature boxes assembled in this way contains feature
boxes in three zones. First there is a source zone (Z1), consisting
of feature boxes subscribed to by the source address of the chain
and applicable to any call in which that subscribing address is the
source address. Second there is a network zone (Z2), consisting of
feature boxes whose presence is required by the network. Third is a
target zone (Z3), consisting of feature boxes subscribed to by the
target address of the chain and applicable to any call in which
that subscribing address is the target address. In each zone, the
order of feature boxes is determined by a fixed precedence
order.
Before routing to the first feature box in a zone, a DFC router
constructs a routing list of feature box types for the zone, and
inserts it into the setup message. When a router routes a setup
message to a feature box, it removes the type of that feature box
from the head of the routing list. The list is copied by the
feature box into the setup message of the continuation call, so
that the router routes the continuation call to the type of feature
box that is now the head of the list. When the routing list is
exhausted, the zone is complete. In this way a chain of feature
boxes with all of the prescribed box types, in the prescribed
order, is assembled.
The implementations of DFC described in various relevant portions
of co-pending U.S. Pat. Nos. 6,404,878 and 6,160,883 are hereby
incorporated by reference. As is described in more detail therein,
new call features may be freely added to or modified within a DFC
environment, and feature specification data (including zone
assignment and priority) for new call features may be readily
integrated.
Turning now to FIG. 1, there is depicted an Extended Communications
Layered on Internet Protocol Synthesis Environment (ECLIPSE) domain
200 for providing distributed, modularized and compositional
communication services in an IP network 100, which may be any type
of communication and/or computer network including physical and/or
wireless transmission media, and hardware that includes
microprocessors and memory for storing processing instructions to
execute the functions described herein. The ECLIPSE domain 200
within IP network 100 allows for rapid deployment and integration
of call features, third party services, user customization and
communication across multiple domains and nodes in an SIP
environment. The management of feature interaction among multiple
ECLIPSE domains 200 can be accomplished with standard networking
topologies.
The ECLIPSE domain 200 offers finite state machine (FSM)
abstraction and a domain-specific language for call feature
specification and development. The ECLIPSE domain 200 may be
operable with a number of third party communication protocols, such
as H.323 and the AMERICA ONLINE TOC protocol. The ECLIPSE domain
200 may further provide an interface to media resources for
accomplishing media processing functions, a flexible provisioning
system and an HTTP interface for administrative and user
self-provisioning functions, fault management, and database
support. ECLIPSE operates with a media abstraction communications
layer for allowing distributed media control by feature boxes. Each
of these functionalities will be described in more detail
commencing with FIG. 2 below.
The ECLIPSE domain 200 may communicate with a plurality of SIP UAs
102, 103 directly and via a plurality of SIP proxy servers 104 and
network routers 106. Although a limited number of components of IP
network 100 are shown in FIG. 1, it should be readily appreciated
that any number of such components may be included. In addition,
any component of IP network 100 may communicate directly with any
other component or via a variety of available intermediary
components as is well known in the art. The ECLIPSE domain 200 may
handle a large number of simultaneous and sequential call requests,
on the scale of a national or global network of users, and must be
able to differentiate call requests based on message header
information. In addition, it should be recognized that, although
SIP UAs 102, 103 are shown as call endpoints in FIG. 1. Additional
client TIs may also be included as endpoints of a call over the IP
network 100.
The ECLIPSE domain 200 operates with a message structure similar to
SIP in which a plurality of headers and fields are used to convey
relevant communications information. The ECLIPSE domain 200 should
recognize all standard SIP headers, such as via, record-route and
route, for purposes of routing a call request.
The ECLIPSE domain 200 performs the general functions of a stateful
SIP proxy server in many regards. Like a standard featureless SIP
proxy server, the ECLIPSE domain 200 should preserve the values of
all SIP header information received within a DFC signaling message.
Feature boxes within the ECLIPSE domain 200, more particularly
described below with respect to FIG. 2, may be assigned to handle
any feature related information.
Previously in SIP, an SIP proxy server would generally never route
the media flow of a call through it. This is because for
featureless calls, it is more efficient to have the media flow go
directly between two call endpoints, such as caller and callee TIs
or SIP UAs 102, 103. In a featureless call in the present
disclosure, when a new call feature is invoked that later requires
control of the media flow, the ECLIPSE domain 200 may attain
control of the media flow using RE-INVITE requests, described later
below.
In both prior SIP domains and the present disclosure, domain
endpoints perform identical functions and generate like responses
regarding the media flow. However, in a call with assigned call
features, the ECLIPSE domain 200 may need to control the media flow
related to such features and so will direct the media flow to be
routed through it.
Similar to previously described SIP UAs that receive media flow for
a call, the ECLIPSE domain 200 is also able to initiate and
terminate a call. For example, the ECLIPSE domain 200 can setup a
conference and send INVITE requests to multiple parties to join the
conference. Thus, the ECLIPSE domain 200 incorporates certain
functionality of existing SIP UAs.
Turning now to FIG. 2, various components of the ECLIPSE domain 200
will be described with respect to particular embodiments, which
should not be viewed as the only configuration in which to
implement the present invention. In certain embodiments, the
ECLIPSE domain 200 includes a plurality of SIP TIs 202 and 203, one
or more routers 204, a database 208 and a plurality of feature
boxes 210, 220 and 230, that may be prioritized first by zone (Z1,
Z2 then Z3) and having a predetermined priority within their
respective zones that allow for feature interaction without
interference. Feature boxes 210-230 may be embodied in separate
physical components or may be logical components within a single
network telecommunications server. The feature boxes 210-230 are
each assigned to a call handled by ECLIPSE domain 200 and provide
the call features required for the call by caller and callee
users.
Calls between feature boxes 210-230 and the SIP TIs 202, 203 are
featureless calls. A variety of feature boxes may be provided
within the system to accommodate a vast number of call features.
Furthermore, a feature box can influence the routing dynamically to
alter the feature boxes assembled into the configuration for a
session. For example, if a feature box in a target zone continues a
call but changes its target address, the DFC router will stop
routing to feature boxes of the previous target address, and begin
to route to target-zone feature boxes of the new target
address.
In this way new and third-party call features are readily
accommodated by providing new feature boxes to the ECLIPSE domain
and specifying their place in the precedence order. This allows the
new features to be used with existing call features, and no
additional effort is required to integrate them. The ability to
assemble call features dynamically and integrate them automatically
thus differs from existing SIP proxy servers, which have no such
capabilities. Feature boxes may be provided to handle registration,
authentication, and the handling of particular SIP response codes,
and to provide other call features such as the follow-me feature
described below with respect to FIG. 3. SIP TI 202 and 203 act as
endpoints for the ECLIPSE domain 200 for sending and receiving
signaling messages over IP telecommunications network 100. Since
line interfaces and trunk interfaces perform similar functions, the
SIP TI 202/203 may be composed of any number of SIP TIs and or LIs.
SIP TIs 202 and 203 may be logically separated components or may be
components of an integrated device. For an ECLIPSE domain 200 with
features, SIP TI 202 and 203 may translate incoming SIP requests to
DFC signaling messages so that they may be understood by feature
boxes 210-230
The original SIP message may be contained as a parameter in the DFC
message that the SIP TI 202 issues. An SIP-aware feature box (not
shown) may examine the original SIP message and perform SIP
specific processing.
Router 204 is provided to route messages based on the source,
target and dialed addresses received in the signaling message. Any
number of routers 204 may be provided to handle appropriate call
volumes. In the cases of featureless calls or the like, media flow
may be routed directly between SIP UAs 102 via media switches 206
and 207 that route and forward media flow for a call, generally as
Real Time Protocol (RTP) packets.
The database 208 is provided within the ECLIPSE domain 200 to store
user information such as provisioned users, call features to be
applied to calls to and from users, and a user's preferred priority
of addresses to which a call to the user is to be redirected.
The ECLIPSE domain 200 handles all registration requests for
provisioning new users, or changing user preferences, thereby
functioning in part as existing SIP registrars. In certain
embodiments, registration may be handled by SIP TIs 202, 203 or by
separate TIs or LIs dedicated to registration requests (not
shown).
The ECLIPSE domain 200 should also be able to distinguish third
party registrations of users, as is common in telecommunications
services, from those performed directly by subscriber-users based
on received user information. For example, a registration request
in which the FROM header contains a uniform resource identifier
(URI) of a third party performing the registration, and in which
the TO address is different from the FROM header (in that it
contains not the same URI but the URI of the subscriber), will be
recognized as a third party registration.
The ECLIPSE domain 200 also incorporates many of the functions of
previous SIP B2BUAs, and may be thought of as an inbound and
outbound SIP UA concatenated together and communicating with each
other and feature boxes 210-230 via DFC routing signals. This
enables the ECLIPSE domain 200 to provide flexibility for call
features that require media flow control. Establishing B2BUA
functions within the ECLIPSE domain 200 also simplifies the
complexity of the call endpoints 202, 203 of the ECLIPSE domain
200. For example, in a call involving a mobile device, such as a
cellular telephone, one endpoint of the ECLIPSE domain 200 may
receive a change in the IP address of the call without notifying
the other endpoint.
Treating the ECLIPSE domain as a B2BUA may, in certain embodiments,
require the alteration of some SIP message headers when the domain
200 relays a call signal, including headers for feature-related
information. For example, the contact header in an incoming invite
call request may be changed to the address of the domain 200 when
the call request is forwarded. Alternatively, the address may be
left unchanged, but instead, host and port designations may be
altered.
User registration is accomplished by the ECLIPSE domain 200 as
follows. Registration requests are handled by SIP registration
feature boxes 210 that are prioritized in the source zone (Z1). A
registered user can change his call contact information by sending
subsequent REGISTER requests to the ECLIPSE domain 200. The SIP
registration feature box stores the received contact information in
the database 208 for later use by a follow me feature box in Z3.
When a SIP registration request arrives at SIP TI 202 or 203, it
will transmit a setup message to the router 204 with the TO header
as the source address, and empty target address and dialed string
fields, thereby invoking only source zone feature boxes such as the
SIP registration feature box. The TO header specified in the setup
or user-status message must be a provisioned address.
Another type of call request handled by the ECLIPSE domain 200 is
the INVITE request for establishing a call.
TABLE-US-00002 FIELD in ECLIPSE HEADER in SIP Setup Message INVITE
message Source Address FROM Target address REQUEST URI Dialed
string TO
When an INVITE is received by inbound SIP TI 202, it formulates a
setup message including the source address of the request, the
target address of the request and the dialed string or dialed
number for the call. This setup message is then routed to the
router to 204 that assigns appropriate feature boxes 210-230 for
establishing call features. Any SDP information in the INVITE
message will be sent to the media switch after call setup. The call
is ultimately routed to outbound SIP TI 203 so that the INVITE
request can be routed to the appropriate contact address.
For incoming INVITE requests, all media specified by a caller may
be indicated in the content of the message and will be put into the
media choice field of the setup message by inbound SIP TI 202. The
outgoing INVITE request to an invited party may be handled in two
manners. A first option is that the outbound SIP TI 203 can send an
INVITE request to the invited party with the media chosen by the
caller. The callee may accept none, part, or all of the media
choices based on the callee's capabilities. One or more accept or
reject messages corresponding to the media choices may be
transmitted back to the caller.
The other option is that the outbound SIP TI 203 transmits the
INVITE request with no SDP to the callee. Upon receiving a
response, it can then send a later avail message or the like with
the media choices of the callee. The caller then selects the
preferred media that is available and communicates the media choice
in the SDP of an acknowledge (ACK) message or the like. This allows
the caller to know the callee's capability before making a media
choice without having to check the parameters of the contact
address registered by the callee.
The ECLIPSE domain 200 may receive RE-INVITE call requests for a
number of reasons, and will have a predetermined response based on
the reason for the RE-INVITE. For example, where the remote user
has changed its IP address (i.e. in a mobile phone system), the
ECLIPSE domain 200 will not notify the SIP UA of the other parties
to the call. The ECLIPSE domain 200 may be adapted to either notify
or not notify invited parties if the codec is changed. The ECLIPSE
domain 200 will, however, notify invited parties if media is added
or subtracted regarding a call.
A wide variety of call features may be readily added and
incorporated in the system disclosed herein. One such call feature
is the follow-me call feature in which calls to a callee address
may be automatically redirected to one or more addresses in a
priority established by a user. The priority and listed addresses
may further have specified period of time for which the preference
is valid.
One implementation of the follow-me feature in the ECLIPSE domain
200 is shown in FIG. 3. A user subscribes to the follow-me feature
as a follow-me feature box provided in Z3. When a call request is
received, the inbound SIP TI 202 generates a setup message with the
dialed string as the TO address. The setup message is routed to
follow-me message boxes based on the target address generated from
the dialed string. When the follow-me feature box receives the
setup message, it will get the contact address stored for the
dialed string stored by the database 208. The follow-me feature box
then replaces the contact address in the target field and continues
routing the call to the invited party at the user preferred
address, or will try multiple addresses (such as an H.323 protocol
device 302, a telephone on a PSTN 304, or another user on the
ECLIPSE domain via SIP TI 203) in priority order. The follow me
feature box may be established as two separate feature boxes, a
general follow-me feature box 231 for handling third party
protocols and an SIP follow-me feature box 232 to handle SIP
specific arguments in a signaling message. This separation allows
ready adaptation of a variety of third party protocols.
FIG. 4 displays a call flow diagram for an exemplary successful
call setup in the ECLIPSE domain 200. A caller SIP UA 102 sends an
invite to inbound SIP TI 202, which acknowledges with a status-code
"100" (trying). The "100" response is meant to stop continuous
re-transmissions of INVITE request only and does not need to be
propagated further upstream. A new call message is sent from the
SIP TI 102 to the incoming media switch 206. The SIP TI 202 then
generates a setup message that is, in turn, sent to outbound SIP TI
203, which responds with an upstream ACK and media message. The
inbound SIP TI 102 responds with open audio and open video messages
that are then acknowledged. External channels and lines for the
media flow are also determined and communicated to the SIP stack. A
ready signal is then sent from inbound to outbound SIP TI 203 to
the outbound SIP TI 203 then sands an INVITE message to the
callee's SIP UA 102.
Callee's SIP UA responds with a "180" (wait) status-code, and may
send multiple messages for each open media channel, which are
propagated upstream ultimately to caller's SIP UA 102. After the
callee's SIP UA 102 provides a "200" status-code (OK) to outbound
SIP TI 203, that response is ultimately propagated upstream to
caller's SIP UA 102. "200" messages are acknowledged by both SIP
TIs and SIP UAs that receive them. Between the inbound and outbound
SIP TIs, the "200" message is propagated as an ACCEPT message. If
some media channels in the original INVITE request are not included
in a 200 response, a reject message will be sent upstream for those
media channels.
An exemplary unsuccessful call setup, as shown in FIG. 5, follows
the same initial steps as that described for FIG. 4, up to
transmission of the INVITE request from outbound SIP TI 203.
However, instead of a "200" response the callee's SIP UA 102
transmits a 4xx (request failure), 5xx (server failure) or 6xx
(global failure) status-code. In response to one of these response,
the outbound SIP TI 203 transmits an unavail message upstream,
resulting in a code message being transmitted from inbound SIP TI
202 to caller's SIP UA 102. A teardown message is then transmitted
from inbound to outbound SIP TIs. Both the code and teardown
messages are acknowledged.
Call flows for various exemplary registration processes between a
user and the ECLIPSE domain 200 are depicted in FIG. 6. A
successful registration process 600 is first depicted. The process
600 commences with a REGISTER request sent from a caller's SIP UA
102 to inbound SIP TI 202. A "100" response is sent in
acknowledgement. Setup and upstream acknowledgement messages are
then sent to the SIP registration feature box. In turn, the SIP
registration feature box contacts and queries the database 208. The
current contact addresses and the expiration time of the contacts
may be obtained from the database 208 or an external database. The
database 208 responds with a success message to the SIP
registration feature box, which in turn, sends an avail message
downstream to the inbound SIP TI 202. A "200" message is then sent
to the caller's SIP UA 102. The registration call is then subject
to teardown.
The SIP BYE request is mapped to the DFC teardown message. When an
inbound or outbound SIP TI receives a BYE request, it will
acknowledge with a "200" response and then send the teardown
message to the outbound SIP TI. When the outbound SIP TI receives
the teardown message, it will send the BYE message to the remote
SIP UA 102.
An unsuccessful registration process 602 due to failed
authentication is next displayed, which proceeds in the same manner
as process 600 up to the response to the query. If the database 208
can not authenticate a provisioned user (for example, when
provisioning data is out of date or further authentication
information is required), it responds to the query with an
authentication-failed message to the SIP registration feature box.
The SIP registration feature box generates an unavail message which
is propagated upstream to inbound SIP TI 202. The inbound SIP TI
202 in turn, sends a "401" (unauthorized) status-code to the
caller's SIP UA 102 with an appropriate challenge in the
www-authentication header of the response, followed by a teardown
of the registration call.
As shown in process 604, a "404" (not found) response could be sent
in place of status-code "401" used in process 602 when a requested
user instead can not be found in the system.
In addition to these processes, the ECLIPSE domain 200 is operative
to respond to other status-code messages. For example, in a process
where status code "183" (session progress) is received, the "183"
response may be handled in the manner described for the "180"
response above. In addition, 3xx feature boxes maybe provided to
handle all 3xx requests requiring redirection of a call. Such 3xx
feature boxes may try to handle redirection of a call where
redirection preferences have been specified by a user. Otherwise,
it may simply propagate 3xx responses upstream for handling by the
domain endpoints.
The ECLIPSE domain 200 is also operative to incorporate other SIP
message requests, such as OPTIONS (for exchanging media
capabilities of SIP components), INFO (for carrying application
level information along the SIP signaling path), REFER (for
performing call transfers)SUBSCRIBE and NOTIFY (to provide user
presence information), MESSAGE (for instant messaging) and DO (for
network appliance control in SIP).
The SIP CANCEL message may be handled in the same manner as the SIP
BYE request described in the proceeding descriptions. The CANCEL
message is mapped to the DFC teardown message by SIP TIs. CANCEL
differs from the BYE request in that it is used before the call is
established. When an SIP TI receives a cancel request for an
existing call setup session, it will send a "200: (OK) response
back followed by a "487" (request terminated) for the original
INVITE request. It will then send a teardown message to the
outbound SIP TI and send the CANCEL request to the caller's SIP UA.
If a final response such as the "200" (OK) status-code message is
sent by the remote SIP UA prior to the CANCEL request, the SIP TI
will instead send the BYE request. If other final responses are
sent by the remote SIP UA (such as 4xx, 5xx and 6xx) the SIP TI
will not send a response.
Although the best methodologies of the invention have been
particularly described in the foregoing disclosure, it is to be
understood that such descriptions have been provided for purposes
of illustration only, and that other variations both in form and in
detail can be made thereupon by those skilled in the art without
departing from the spirit and scope of the present invention, which
is defined first and foremost by the appended claims.
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